The ability to control sound radiation patterns in entertainment, gaming, communication and personal messaging is becoming an important differentiating feature in many commercial products. A common aim in these systems is to create a highly-directional sound field to targeted audiences by forming a tune-in zone (or personal audio) for a group of people. There are several ways to generate the directional sound field. These include (i) using a sound dome that projects sound to a convex surface to focus sound waves to the listeners below the sound dome; (ii) using a loudspeaker array with the phase-amplitude differences between different loudspeakers adjusted to spatially steer an audible sound beam in a horizontal plane; and (iii) modulating an audible sound signal onto an ultrasonic carrier signal and projecting the modulated signal via special types of ultrasonic emitters to generate a parametric array through the air in such a way that audible sound can travel in a column of sound beam. Loudspeakers generating the directional sound field using (iii) are commonly called parametric (or ultrasonic) loudspeakers. The parametric loudspeaker is based on a nonlinear acoustics property (known as the parametric array effect in air) that uses ultrasound signal to carry the audible sound signal in a tight beam, just like an audio spotlight.
When using a loudspeaker array (as described in (ii)) to steer an audible sound beam at low frequencies, for example, at frequencies less than 200 Hz, the dimension of the loudspeaker array must be significantly greater than the audio wavelength in order to achieve a good directivity. Usually, this means that the dimension of the loudspeaker array must be more than a meter in diameter. This approach of creating a focused sound beam hence incurs a high cost since a large loudspeaker array is required. In contrast, a parametric loudspeaker (as described in (iii)) is able to generate a highly-directional sound beam for a low-frequency sound wave whose wavelength is much larger than the loudspeaker diameter. This is because the small-sized ultrasonic emitter in the parametric loudspeaker is able to produce a highly-directional sound beam without using a vibrating cone as opposed to conventional loudspeakers.
FIG. 1 illustrates a parametric loudspeaker according to the prior art. In the parametric loudspeaker, an ultrasonic carrier signal is first modulated by a modulating input signal which is in the form of an audible sound signal. Preprocessing and modulation units are used to generate the modulated signal. The modulated signal is then passed to an amplifier that drives the ultrasonic emitter to project the modulated signal through a transmission medium (usually air). As the modulated signal is radiated into the transmission medium, it interacts with the transmission medium and self-demodulates to generate a tight column of audible signal. An audible sound beam is thus generated in the transmission medium through a column of virtual audible sources as shown in FIG. 1. This column of virtual audible sources forms an end-fire array of audible sources (referred to as a parametric array) that add up in phase along the propagation axis.
The Berktay far-field model is widely used to approximate the nonlinear sound propagation by the parametric loudspeaker through the transmission medium. This model uses an expression as shown in Equation (1) to predict the far field array response of the parametric loudspeaker. According to Equation (1), the demodulated signal (or audible difference frequency) pressure p2(t) along the axis of propagation is proportional to the second time-derivative of the square of the envelope of the modulated signal when amplitude modulation is used. In Equation (1), β is the coefficient of nonlinearity, P0 is the primary wave pressure, a is the radius of the ultrasonic emitter, ρo is the density of the transmission medium, c0 is the small signal sound speed, z is the axial distance from the ultrasonic emitter, α0 is the attenuation coefficient at the source frequency and E(t) is the envelope of the modulated signal.
                                                                                          p                  2                                ⁡                                  (                  t                  )                                            ≈                            ⁢                                                                    β                    ⁢                                                                                  ⁢                                          P                      0                      2                                        ⁢                                          a                      2                                                                            16                    ⁢                                          ρ                      0                                        ⁢                                          c                      0                      4                                        ⁢                    z                    ⁢                                                                                  ⁢                                          α                      0                                                                      ⁢                                                      ⅆ                    2                                                        ⅆ                                          t                      2                                                                      ⁢                                                      E                    2                                    ⁡                                      (                    t                    )                                                                                                                          ∝                            ⁢                                                                    ⅆ                    2                                                        ⅆ                                          τ                      2                                                                      ⁢                                                      E                    2                                    ⁡                                      (                    t                    )                                                                                                          (        1        )            
As shown in Equation (1), the nonlinear sound propagation results in a distortion in the demodulated signal. This in turn results in a distortion in the audible signal generated by the parametric loudspeaker, hence affecting the performance of the parametric loudspeaker. Furthermore, the current parametric loudspeaker technology is severely limited by the technological constraints of ultrasonic emitters. One such technological constraint is the small usable low-frequency bandwidth of the ultrasonic emitters.
Digital signal processing techniques have previously been proposed to overcome the technological limitations of the parametric loudspeaker technology. These techniques usually involve pre-processing algorithms which can be programmed in a digital signal processor to enhance, equalize and compensate for any distortion in the audio quality of the signal before sending the processed signal to the ultrasonic emitter. Examples of such techniques are described below.
FIG. 2 shows an adaptive parametric loudspeaker system 200 proposed in U.S. patent application Ser. No. 11/558,489 “Ultra directional speaker system and signal processing method thereof” (hereinafter, Kyungmin). Kyungmin proposes adaptively applying pre-distortion compensation to the modulating signal x(t) (i.e. the input audible signal). Furthermore, instead of using a double sided amplitude modulation (DSBAM) scheme typically used in parametric loudspeaker systems, Kyungmin proposes the use of vestigial sideband modulation (VSB) to overcome the non-ideal filtering of one of the sidebands in single sideband (SSB) modulation.
As shown in FIG. 2, the adaptive parametric loudspeaker system 200 comprises 1st and 2nd envelop calculators 202, 204 which calculate the envelops E1(t) and E2(t) respectively. These envelop calculators 202, 204 are injected with signals at the baseband. The adaptive parametric loudspeaker system 200 also comprises a square root operator 206 which computes the “ideal” envelop √{square root over (E1(t))} predicted using the Berktay's approximation (i.e. Equation (1)). The difference between √{square root over (E1(t))} and E2(t) is then used to train the pre-distortion adaptive filter 208 using the least mean square (LMS) scheme. The coefficients am of the adaptive filter 208 are obtained using Equations (2) and (3) as follows wherein β is an adaptive coefficient.a′m(t)=−2(√{square root over (E1(t))}−E2(t))×(t−m)  (2)am(t+1)=am(t)+βa′m(t)  (3)
The output 40 of the adaptive filter 208 is shown in Equation (4) as follows.
                                          x            ′                    ⁡                      (            t            )                          =                              ∑                          m              =              0                                      N              -              1                                ⁢                                                    a                m                            ⁡                              (                t                )                                      ×                          (                              t                -                m                            )                                                          (        4        )            
A pre-processing technique is also proposed in U.S. Pat. No. 6,584,205 (hereinafter, Croft) to improve the performance of parametric loudspeakers. FIG. 3 illustrates a parametric loudspeaker system 300 proposed in Croft. Croft proposed the use of SSB modulation as it offers the same ideal linearity as characterized by square rooting a pre-processed DSBAM modulated signal. Croft further proposed compensating for the distortion inherent in SSB signals using a multi-order distortion compensator. The multi-order distortion compensator comprises a cascade of distortion compensators (Distortion compensator 0 . . . N−1 as shown in FIG. 3) whereby a pre-distorted signal (for example, x1(t)) from one distortion compensator is used as the input to the next distortion compensator in the cascade and so on, until the desired order is reached. Each distortion compensator of Croft comprises a SSB modulator 302 which employs a conventional SSB modulation technique. Similar to Kyungmin, the non-linear models 304 shown in FIG. 3 are based on Berktay's approximation (i.e. Equation (1)) and the system 300 proposed in Croft is based on a feed forward structure found in the multi-order distortion compensator.